Hello,
I'm sorry for this basic question, but I have SER running on my LAN with
(RFC1918) addresses with UAs on the same network.
Then there are UAs registering from behind NAT to this SER on the LAN
network.
(SER is not listening on any public IP and it's just used in the private
network)
I came across this issue. If a UA on the LAN network is registering to
SER the logic of ser.cfg recognize the UA is behind NAT
after nat_uac_test.
If I want to introduce NAT traversal with RTPproxy for this private
network nat_uac_test will always return that the UA
is behind NAT because it has RFC1918 IP address. The RTPproxy will be
used at all times for UAs on the network or even I introduce STUN for
NATed UAs
I know this is not a common usage of SER because it is invented to be
used on the global internet but I was just wondering is there any
possibility
not to use RTPproxy at all times for UAs on the private network?
Maybe this is a silly question but it would help me for my testing
purposes with SER where I will not have to use my setup with usage of
public IPs.
Thank you for any thoughts.
Ladislav
I am trying to use FIFO to manage permanent contacts. I can create
contacts now, but can't figure out what value to specify for the
expiry.
A positive value represents a real expiry in seconds. Negative
numbers, "Never", and "Permanent" all result in "400 Invalid expires
format". A zero results in "500 Error while adding contact".
The information at
http://www.openser.org/dokuwiki/doku.php?id=fifo_relay#ul_add is more
of an example. It doesn't really explain the arguments. Does anyone
know how to specify that a contact is permanent? The command
"openserctl ul add" is able to create permanent contacts, so it must
be possible.
Doug
--
Doug Meredith (doug.meredith(a)systemguard.com)
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
I am using OpenSER 1.0. When I send the following to the FIFO:
-----------------
:ul_add:ans
location
81(a)dev.skyridge.com
sip:34@sirloin.skyridge.com
3600
1
-----------------------
I get the response "400 ul_add: replicate expected". I have done a
google search without finding anything helpful. Can anyone explain
what this means?
Doug
--
Doug Meredith (doug.meredith(a)systemguard.com)
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
On my test network I have set up a dispatcher server that points to two
proxy servers that hit an external database.
I have a simple route script on the dispatcher that should be routing
requests to the two proxy servers.
dispatcher .51
caller .99
recipient .100
proxy1 .52
proxy2 .53
I'm making call requests from .99
to .100, with .51 used as the proxy server.
The problem I'm getting, is that the INVITE is being sent directly to
the recipient without passing through the proxys, the dispatcher is
passing the message directly. At least i think that is what is
happening.
Here is the sip INVITE:
INVITE sip:rick@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.99:5060;rport;branch=z9hG4bK837723126
From: <sip:rick@192.168.0.51>;tag=679508646
To: <sip:rick@192.168.0.100>
Call-ID: 125992616(a)192.168.0.99
CSeq: 20 INVITE
Contact: <sip:rick@192.168.0.99:5060>
Max-Forwards: 5
User-Agent: Linphone-1.0.1/eXosip
Subject: Phone call
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE,
NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 352
So it seems to me that the invite request is being sent to .51 and it is
forwarding directly to .100 without dispatching the message to .52
or .53
here is my dispatcher config:
# $Id: dispatcher.cfg,v 1.1.1.1 2005/06/13 16:47:37 bogdan_iancu Exp $
# sample config file for dispatcher module
#
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
log_facility=LOG_LOCAL0
children=2
check_via=no # (cmd. line: -v)
dns=off # (cmd. line: -r)
rev_dns=off # (cmd. line: -R)
port=5060
fifo="/tmp/openser_fifo"
# for more info: sip_router -h
# ------------------ module loading ----------------------------------
loadmodule "//lib/openser/modules/xlog.so"
loadmodule "/usr/lib/openser/modules/maxfwd.so"
loadmodule "/usr/lib/openser/modules/sl.so"
loadmodule "/usr/lib/openser/modules/dispatcher.so"
# ----------------- setting module-specific parameters ---------------
# -- dispatcher params --
modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list")
# modparam("dispatcher", "force_dst", 1)
route{
if ( !mf_process_maxfwd_header("10") )
{
sl_send_reply("483","To Many Hops");
drop();
};
ds_select_dst("2", "2");
forward(uri:host,uri:port);
# t_relay();
}
Okay... I'm a little stumped on what's going on here, so I'll start with a
general explanation before I bother posting my (likely hard to read) ser.cfg.
I'm having a weird issue when calling from X-Lite to a SNOM 320 phone (I
haven't tried the SNOM 190 with a different firmware yet, but I don't see that
it would make much difference).
I have a failure route which handles voicemail and such, and I have the invite
timer set to 27 seconds (roughly 4.5 standard US rings). Now, when I call from
X-Lite to X-Lite, or from Leadtek 8051 to X-Lite, or from SNOM to X-lite, or
from SNOM to SNOM, or from Leadtek to Leadtek, etc, etc... everything works
EXACTLY as it's supposed to. I have log messages telling me when the failure
route is being entered, and it hits it at right on 27 seconds.
However, when I call TO the SNOM phone from any other device besides another
SNOM phone, I never hit the failure route until a 408 timeout at 58 seconds.
Now, I can't think of a reason for a UA to overwrite the invite timer on SER
or even how such overall weirdness would occur, but it's 58 seconds every time
like clockwork.. error 408. At that point, it enters the failure route and
does its processing (forwarding the call to voicemail as far as SER is
concerned but not as far as the UA is concerned... which just receives an
error message).
Any ideas?
Also... when I dial SNOM to SNOM, the SIP method I get in the ACC is PRACK (no
INVITE). Can someone fill me in on what a PRACK is?
Is there something I should be looking at to figure out why calling one UA
behaves differently than other UAs?
N.
Hi list,
I am so new to list SER.
I just tried to
I want to authenticate SIP users via RADIUS and want accounting with radius.
also if possible, ask route to radius also.
but I did not manage it ;(
any simple sample? I read documents etc. no good sample found.
--
gMail : google mail
Try it! It is the best
http://www.gmail.com
Hello,
I have setup openser and I have a request to incorporate an
unconditional forward feature which can be turned on by
dialing a "*21+ number" and can be turned of by dialing #21#.
Can anyone provide me with an example piece of .cfg file that can do
this ?
best regards
Paul
Hi,
I have a question more in a general aspect.
Since the draft-levy-sip-diversion-08 has expired, should a Proxy strip
such headers?
According to RFC3261,the UAS MUST ignore headers-fields not defined in
the specification or any supported extension...
I've realized that some systems need Diversion headers for voicemail (I
know Asterisk doesn't, but Cisco Unity does)
Any comments?
--
Helge Waastad
Senior Engineer
Systemavdelingen
Smartnet
Hello all,
Is there anything that needs to be done to the ser cfg file to support
STUN? If so, could you please share that config snippet?
Thanks in advance.
--
Regards,
Jan Henkins
Cell: +27-84-951-4334
---------- Forwarded message ----------
From: Voipers Portugal <voipers(a)gmail.com>
Date: Dec 2, 2005 9:08 AM
Subject: Re: [Serusers] NAT transversal
To: jan(a)henkins.za.net
I think you should start by reading the nathelper module. There you'll find
the information that you will need. If you want, try the OnSip config file
for NAT, it's quite self explaning, and it works.
Jose Simoes
On 12/2/05, Jan Henkins <jan(a)henkins.za.net> wrote:
>
> Hi,
>
> I'm trying to set up ser (version 0.9.4) on a server that is not behind
> any form of NAT. I need to get to this ser server from behind NAT
> firewalls (ADSL). To test I'm using a variety of software (linphone,
> Twinkle etc.), and I also have a Grandstream Budgetone 100 that I'm
> using. I tested the Grandstream phone successfully on my FWD account
> with STUN, however try as I might I can not log on to my server. What is
> the best way to accommodate this type of setup? At the moment I need a
> simple nudge into a direction so that I can research it properly.
>
> Thanks in advance!
>
> --
> Regards,
> Jan Henkins
> Cell: +27-84-951-4334
>
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>
